1. Field of the Invention
The present invention relates to an apparatus and method for testing a voice over internet protocol (VoIP) network.
2. Description of Related Art
Voice over Internet Protocol (VoIP) is the newest technology available for making telephone calls. Unlike existing “copper” technology, where each customer has dedicated wires directly from their home or business to a central office or a local cabinet, VoIP relies on transmitting messages over a high-speed network, such as a coaxial cable network utilizing a cable modem, such as a DOCSIS cable modem, in the cable TV industry or a conventional telephony network utilizing a Digital Subscriber Line (DSL) modem by existing telephony providers.
Utilizing traditional copper-based telephony, phone calls are established between two telephones using analog signaling methods, called “Ear & Mouth” signaling, or E&M. This is performed by changing voltages, polarities, and transmitting DTMF (dual-tone multi-frequency) tones, which are interpreted directly by switch equipment in the central office. For example, to indicate a phone is off-hook, the E signal is connected to ground.
When using VoIP technology, since the physical wires and/or the analog signaling methods currently used by the copper technology are no longer present, the traditional analog signals must be converted into digital packets transmitted over the corresponding high-speed network. This conversion is done at customer premise equipment (CPE) sides of the network by devices known as multimedia terminal adapters (MTAs) and embedded MTAs (EMTAs). MTAs and EMTAs are also known as “endpoints.” Between the endpoints of a VoIP telephone call, the digital packets are interpreted by VoIP switch equipment, such as a Call Management System (CMS), which is the equivalent of the CLASS 5 (Custom Local Area Signaling Services) switch for analog telephones residing in the central office, which is configured to properly connect the endpoints of a VoIP telephone call.
Various new, “message-based” signaling methods have been created to enable VoIP phone calls to be established. The two most popular of these methods are the Media Gateway Control Protocol (MGCP) and the Session Initiation Protocol (SIP). Another method called Network Call Signaling (NCS) has been developed specifically for the Hybrid Fiber/Coax (HFC) business. NCS is based on MGCP.
MGCP and NCS signaling methods both provide the ability for the CMS to put the endpoint into two special modes, namely, a network loopback mode and network continuity mode. Both of these modes are used for testing and diagnostic purposes at the network level to ensure that network connectivity is present and that at least one of the endpoints is performing codec (code & decode) translation properly.
A problem with the MGCP and the NCS signaling methods is that neither of these methods enable an originating endpoint of a phone call to cause a called or destination endpoint of the phone call to enter into either the network loopback mode or the network continuity mode for a particular phone call.
It would, therefore, be desirable to provide a method and apparatus that overcomes the above problems and others. Still other problems that the present invention overcomes will be apparent to those of ordinary skill in the art upon reading and understanding the following detailed description.